Hi!
We use the ancient FreePBX 2.5.2.6 and asterisk 1.4.
In which .conf do I find the current audio-codecs?
Where to enable jitterbuffer?
The usual asterisk configs are auto-generated, but I don’t know with which configs I’ve to work then?
;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat’ing when going
;through a firewall. For nat’ing you’d need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf
The quote above is from the sip.conf. As it’s written the sip_general_custom.conf handles the jitterbuffer, but this config is commented out by the # before. So, does it even make sense to write in the sip_general_custom.conf?
Thx in advance 